Telephone systems and IP-based mobile systems provide for communication between telephone units, including the use of mobile user equipment on wireless communication networks. Among other elements, the transition of connectivity from one communication network to another network usually involves a transition of communication services. The user equipment, source network, and target network may be called different names depending on the nomenclature used and the base technology used in the particular network configurations or communication systems.
An IP-based mobile system includes at least one mobile node on a wireless communication system. A “mobile node” is sometimes referred to as user equipment, mobile unit, mobile terminal, mobile device, or similar names depending on the nomenclature adopted by particular system providers. The various components on the system may be called different names depending on the nomenclature used on any particular network configuration or communication system.
For instance, “mobile node” or “user equipment” encompasses PC's having cabled (e.g., telephone line (“twisted pair”), Ethernet cable, optical cable, and so on) connectivity to the wireless network, as well as wireless connectivity directly to the cellular network, as can be experienced by various makes and models of mobile terminals (“cell phones”) having various features and functionality, such as Internet access, e-mail, messaging services, and the like. The term “mobile node” also includes a mobile communication unit (e.g., mobile terminal, “smart phones,” nomadic devices such as laptop PCs with wireless connectivity).
According to the RFC 3261, June 2002, SIP is an application layer protocol for establishing, terminating and modifying multimedia sessions. “SIP: Session Initiation Protocol”, RFC 3261, June 2002. SIP communications are typically carried over IP protocols and IP networks. Telephone calls are considered a type of multimedia sessions where just audio is exchanged.
ISUP is a level 4 protocol used on traditional switched telephone (SS7) networks. ISUP communications are typically carried over MTP, although such communications can also run over an IP network. “Application of the ISDN user part of CCITT signaling system No. 7 for international ISDN interconnections,” ITU-T Q.767 recommendation, February 1991. ISUP is used for controlling telephone calls and for maintenance of the network blocking circuits, resetting circuits, and other network equipment.
There are several flavors of ISUP. ITU-T Q.767 International ISUP is used through this document; some differences with ANSI ISUP (“Signaling System No. 7; ISDN User Part” T1.113-1995 ANSI, January 1995) and TTC ISUP are outlined. ISUP Q.767 is the least complex of all the ISUP flavors. Due to the small number of fields that map directly from ISUP to SIP, the signaling differences between Q.767 and specific national variants of ISUP will generally have little to no impact on the mapping. It should be noted that the ITU-T has not substantially standarized practices for Local Number Portability since portability tends to be grounded in national numbering plan practices, and that consequently LNP must be described on a virtually per-nation basis.
The manner of mapping between two signaling protocols: the Session Initiation Protocol (SIP) and the ISDN User Part (ISUP) of SS7 is described in Q.1912.5 Profile C and, RFC 3398, December 2002, which focuses on the translation of ISUP messages into SIP messages, and the mapping of ISUP parameters into SIP headers, and vice versa.
A module performing the mapping between these two protocols is usually referred to as Media Gateway Controller (MGC), although the terms ‘softswitch’ or ‘call agent’ are also sometimes used. An MGC has logical interfaces facing both networks, the network carrying ISUP and the network carrying SIP. The MGC also has some capabilities for controlling the voice path. There is typically a Media Gateway (MG) with E1/T1 trunking interfaces (voice from PSTN) and with IP interfaces (VoIP). The MGC and the MG can be merged together in one physical box or kept separate.
The Mapping between SIP headers and ISUP parameters in RFC 3398, December 2002, focuses largely on the mapping between the parameters found in the ISUP Initial Address Message (IAM) and the headers associated with the SIP INVITE message. Both of the IAM and SIP INVITE messages are used in their respective protocols to request the establishment of a call. Once an INVITE has been sent for a particular session, such headers as the To and From field become essentially fixed, and no further translation will be required during subsequent signaling, which is routed in accordance with Via and Route headers. Hence, the problem of parameter-to-header mapping in SIP-T is confined mostly to the IAM and the INVITE. There are other mappings that are important as well such as the mapping of Address Complete Message and Call Progress Messages to SIP 18× response messages. Q.1912.5 defines details on the mapping between ISUP and SIP in various scenarios and configurations.
The population of parameters in the ISUP ACM and REL messages based on SIP status codes is addressed in RFC 3398, December 2002, which also describes when the media path associated with a SIP call is to be initialized, terminated, modified. This RFC does not, however, go into details about how the initialization is performed or which protocols are used for that purpose.
A “loop condition” arises during initialization of a call across the ISUP/SIP boundary interface where a telephone call is routed repeatedly between the ISUP and SIP domains with a non-decrementing hop counter. This condition results in the same telephone call being processed indefinitely. If the “loop condition” is not broken, system resources will continue to be consumed on a telephone call that cannot be connected, instead of dropping the attempted telephone connection. What is needed is a method of breaking this “loop condition” and break the cycle of providing the same (or higher) parameter values between networks at the network boundary for an uncompleted telephone connection.